Hi,
What's the perfered way to launch Asterisk on boot, on say Redhat?
I tried putting this in my rc.local:
modprobe wcfxo
/usr/bin/screen -d -m /usr/sbin/asterisk -vvvcn
But it doesn't work, and I get this in my asterisk message log:
Aug 6 10:24:27 WARNING[16384]: File codec_g729b.c, Line 50
Hi,
I need to know from your experience which is the minimum hardware
configuration (proc./memory) to run * with two X100P cards and around 10
internal IP phones (from which 4 are hardware ones and use G.711, the rest
of them with GSM), with a couple of IAX connections with other Asterisk
PBX'es (
I've searched the archives a bit and have not really come up with
a good answer to my queries.
I have * running on a RH9 box behind a LinkSys NAT box. I can talk
with iConnectHere outbound just fine. I am trying to configure an
inbound Xten softphone from outside. I have that user set as NAT in
I have my new phone mostly working. I do have a couple of residuals
that I cannot find mentioned in the list archives:
1. Is it possible to set the volume in these things? I hope I didn't
miss it, but I've looked in the doc, the FAQ, and the asterisk archives
and don't find anything. The dis
well should be ok if you cvs update now.
Martin
On Wed, 6 Aug 2003, Rhys Hopkins wrote:
> Martin Pycko wrote:
> > You're looking for libncurses-dev and in libpri you can remove -Werror
> > from libpri/Makefile or cvs update libpri (it should be fixed)
> >
>
> Thanks for that - I installed ncurse
Garry,
yes this is possible although it would end up being quite convoluted.
Essentially you could have a cron job that monitors your voicemail directory, or use
the perl manager interface to check the status. Once it has been established that you
have message(s) submit a .call file to dial you
I am trying to record all logins and logouts for posterity. The act of
logging in with AgentCallbackLogin produces the same CDR data as logging
out. Is there some way to distunguish them in CDR? I also noticed the
management interface maintains a Unique ID for each call and lets that
call be tr
On Wed, 2003-08-06 at 16:20, Andy Powell wrote:
> It's just a proxy service like fwd it will work with asterisk... The phones they are
> selling
> with the deal are Grandstreams.
Perhaps that explains why nobody can get to the site to order
Grandstreams right now. :)
Hi
Is it possible to use a sip gateway as a trunk.
If so, how would I do this
David Hindmarsh
- Original Message -
From: "Jamie Carl" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 07, 2003 12:14 PM
Subject: Re: [Asterisk-Users] X-Lite <-> Snom200
> Yes, over a L
Hi Dave,
On Tuesday 05 August 2003 13:53, Dave Wilson wrote:
> Hi all,
>
> I can't seem to find any info on this anywhere on the web, except that BT
> caller ID doesnt use the standard bellcore system in use in the US. So, if
> anyone here in the UK is onlist and using the x100p successfully, plea
Anyone have any luck setting up festival 1.4.3? Can someone share some
input.
bkw
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> out. Is there some way to distunguish them in CDR? I also noticed the
> management interface maintains a Unique ID for each call and lets that
> call be traced throughout its life in the PBX. Can that data be added to
> CDR as well to allow for easier call tracking?
It looks like if you defi
All,
Could one use the standard analoge modem to test Asterisk functionality.
I mean just the phone and line jack OR do I need specific hardware?
Thanks in advance
Jeroen
-
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I don't think that's the same unique id. It changes for each record in
the CDR. I believe the management interface unique id is maintained as
specific to each incoming or 'original' call. Any ideas?
Jim Friedeck
---
Jim Friedeck wrote:
Thanks! I'm trying that now.
Jim
use "extensions reload" CLI command
Martin
On Wed, 6 Aug 2003, Steven J. Sobol wrote:
>
> Is there a way to make * forget that SIP phone
> [EMAIL PROTECTED] is registered? I ask because I have a few
> different PSTN numbers that I use for various reasons, and I can reprogram
> my Grandstream, bu
He should treat the first part as a local extension.
amark
On Thu, 7 Aug 2003, Michael Ulitskiy wrote:
> Hi,
>
> Does anyone know if asterisk can handle 3xx SIP responces?
> I'm trying make it work with redirect server and it looks like
> asterisk isn't going to send another invite, but treats "
> could you send me the exact syntax for rxgain / txgain?
> I think that might help towards my problem
> becuase i'm having to turn the handset volume all the
> way up ..
>
> thanks
> Dave
You can use either a percentage or a number IIRC..
Somthing like..
rxgain=5%
txgain=5%
or
rxgain=0.4
txg
Dan wrote:
add in the [pos] section :
username=pos
and then if it still doesn't work try to comment the lines:
;deny=0.0.0.0/0.0.0.0
;permit=10.1.3.0/255.255.255.0
;defaultip=10.1.3.2
Did all that, even restarted asterisk (instead of reload via CLI), still
no go. Any other idea?
rgds
pos
I am tryin
Steve
I have to say that some listserv's do allow this .. at least he didn't reply to 20
messages with
REMOVE
in them
Andy
On 07/08/2003 at 10:10 Steve Meyers wrote:
>On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
>> unsubscribe
>
>Has anyone ever been on a mailing list where you co
Hi
I don't know if this is a bug or a configuration issue. I have Cisco 7960
phones off of my *.
Voicemail works fine, but I can't get the MWI light to work correctly. I
have the phones deposit the VM in a VM directory that is not the default one
(directory called "sip"). If I push VM files into th
Since the DSP is software driven for the Wildcard product. Is there a
benchmark out that depicts how much processor is utilized on TDM calls
per codec that's being used?
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Thanks, that fixed it.
Now, the problem would be if you wanted to keep voicemail in different
directories for a mulit tenant asterisk server. Anyway of making MWI use a
different directory (on a per user basis)?
Lee Goodman
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To:
On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
> unsubscribe
Has anyone ever been on a mailing list where you could unsubscribe
simply by sending a message with "unsubscribe" in it to the mailing
list? I swear, every list I've been on, people try to do that, but it
doesn't work on any of them
You also appear to have a big problem with your clock...
unless you are from the future.. in which case how are Glaxo stocks doing?
Andy
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> I've taken the liberty to edit your patch, to put back in the
> 'adsi_logo' and the values for adapp and adsec as they are in CVS. As
> far as I can tell those changes have no relation to problem this patch
> solves, they're just local changes to satisfy your local preferences,
> right? I've rem
I've canceled my Vonage service because of the requirement to prefix
every call with a 1.
Vonage has charged me $42 & will refund this when they get my ATA186
back. I'm thinking I should keep the 186 for possible use with Asterisk.
Anyone know if this will work out?
As I understand it, the pr
hi,
i have connected a SNOM 200 to the asterisk. here are my settings,
Codecs
---
Default codec - g.711u/g.711a
Packet size - 20ms
Negotiation - Interoperable
Type - 160
DTMF
Inband - negotiate
Outband - negotiate
Payload Type - 101
when a call comes to the SNOM or when making an out
On Thu, 2003-08-07 at 01:56, Brian Capouch wrote:
> 2. This phone does not act like all my others do when I am talking and a
> call comes in. Instead of the jarring ADSI !!!BOING!!! followed by a
> series of call waiting beeps, instead I get a ringing tone in the
> earpiece which is audible to
Digium is planning to make some Asterisk/Digium t-shirts. We'd like to
have people submit t-shirt designs from which we might select one in
addition to whatever we might come up with on our own.
1) The t-shirt should be primarily for Asterisk but should contain the
Digium logo somewhere, too.
2)
> I have my new phone mostly working. I do have a couple of residuals
> that I cannot find mentioned in the list archives:
>
> 1. Is it possible to set the volume in these things? I hope I didn't
> miss it, but I've looked in the doc, the FAQ, and the asterisk archives
> and don't find anythi
Sorry about taking this OT
How do you go about "you could write a quick LDAP->XML dump with perl"
Please just point me to a howto.
TIA
John Todd wrote:
Nice...so mixing and matching IP and POTS is ok and common then?
Yes.
I do care what the IP phones cost but then again I'm a gadget
fre
Michael Ulitskiy wrote:
Michael,
With all due respect to both of you, it's not related to h.323 driver.
The result is the same whether h.323 channel participates in the
call or it's pure sip-to-sip call.
Did you try it without the ztdummy and zaprtc?
I posted to Mark a couple of patches fixing M
Tones are to short.
Stefano Finetti wrote:
Mark,
I'm now able to send proper DTMF tones checking on the isdn driver and using
"rfc2833" as dtmf mode for sip.conf and phones.
But there is a question that i think only you can check and answer:
Why * often when calling via outside line some number
That answers one question for me. I won't be buying any of those vonage
ATA-186s being advertised on e-bay
Brian Capouch wrote:
John Schmerold wrote:
I've canceled my Vonage service because of the requirement to prefix
every call with a 1.
Vonage has charged me $42 & will refund this when the
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